Cant get company name from!

fenixdemetal

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#1
Hello everyone, i have just configured the hylafax service to incoming faxes, i only have a problem the template e-mail do not show me the {COMPANY_NAME_FROM} and i relly need that data to know who send the faxes with out open it. where is {COMPANY_NAME_FROM} it shows me the phone number again, is this normal ? or there is some way to get that information

i will apreciate any help
Regards
 

danardf

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#2
Did you put a lookup source into incoming calls? (lookup from database, or freepbx phonebook).
 

Patrick_elx

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#3
fenixdemetal said:
Hello everyone, i have just configured the hylafax service to incoming faxes, i only have a problem the template e-mail do not show me the {COMPANY_NAME_FROM} and i relly need that data to know who send the faxes with out open it. where is {COMPANY_NAME_FROM} it shows me the phone number again, is this normal ? or there is some way to get that information

i will apreciate any help
Regards
If you look in your cdr report, is the number displayer with a Caller Id Name? If not, it's because it was not provided to you by your provider.

As said Franck, you maybe want to add a CNAME lookup source (I would refer you to the marvelous CallerID Superfecta additional Freepbx module)
The way freepbx build the dialplan, if no CNAME is provided or found in a lookup source, the CNAME is set as the CID Number.

I'm not sure if Hylafax is using the CNAME as Company_Name_From or if it's using the fax header. But even in that case, was the fax header company name really transmitted, or just printed on the scan before faxing?
 

kongar

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fenixdemetal

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#5
Thanks you guys for the answers, i think my problem is becouse as Patrick_elx saids when i go to cdr report i cant see the company name i only see in source the "USER Context" number I used in the trunk, and this is the same it shows me in the template e-mail (not like i said first it shows me the phone number again, ups!) so i have added some numbers whit the name in the asterisk phonebook on free pbx module, but... what else i have to do the the incoming calls take this changes ?

thanks
 

Patrick_elx

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#6
you need to make sure that CID lookup source in your inbound route is selected to the correct source (in your case internal).

But I would suggest to add the CID Superfecta module that will give you more lookup source options.
 

fenixdemetal

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#7
Hi, i allready added CID Superfecta module to the free PBX, i follow this steps:
http://www.fonicaprojects.com/wiki/inde ... Superfecta
now i use HTTP source on Caller ID Superfecta options, whit a user and password of a new user only with CID superfecta permisions, i have already added some names whit numbers on asterisk phone book (or i have to added some where else ? ), but when i test all this numbers with debug button, It seems to be loading something but i cant see anything. am I missing something ??
I apreciate your help!
thanks
 

Patrick_elx

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#8
fenixdemetal said:
now i use HTTP source on Caller ID Superfecta options, whit a user and password of a new user only with CID superfecta permisions,
That I don't understand.

If you are using Elastix, you don't need to enter anything in the superfecta page regarding http user and password.

You will add as source in superfecta asterisk db and other if you want (country dependant).

You need however to add the line
RewriteCond %{REQUEST_URI} !(/admin/modules/superfecta/*)

at the proper location in /etc/httpd/conf.d/elastix.conf

Then, in your inbound route you select CID superfecta as lookup source.
 

fenixdemetal

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#9
ok, as i said, i follow the steps on: http://www.fonicaprojects.com/wiki/inde ... Superfecta , there I found that

For FreePBX Administrative ACL users ONLY:

Step 4: Using FreePBX, create a new user in Administrator. Assign this user permissions over ONLY the SUPERFECTA module.

Step 5: Use a web browser to access FreePBX on your PBX. Choose Setup, Caller ID Look Up Sources.

The module installer has already created the entry for Caller ID Superfecta. Select if from the list. In the USERNAME and PASSWORD, enter the Username and password you created for this purpose in FreePBX Administrator.
thats what I mean, but you tell me to add as source in superfecta asterisk db, did you mean the internal option in source type on CallerID lookup sources ? well i am not shure but I already modify the /etc/httpd/conf.d/elastix.conf file and test this settings with out good results :( ...
I added some numbers whit names on asterisk phonebook and no where else, for example i am testing for the number identified as 6477134 so i put this number whit the name "test" on asterisk phone book, but it dont show me the name "test" in the cdr report.
only the name and trunk number.

what cant be wrong ?
 

Patrick_elx

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#10
In the Superfecta module page (ie FreePBX, CID Superfecta left column tab)

You select the default scheme for instance,
then in this scheme you have a list of data source name.
You need to choose 'Asterisk Phonebook' and enable it. If this source is not here, click the update box and find it in the drop down box.

In the bottom of the page you can keep CID rules, DID Numbers, Username, Password and CID Prefix URL empty


Now go to the Inbound Route tab in FreePBX and for your routes, make sure that CID lookup source has selected Caller ID Superfecta.

Also if you look in the CDR to see the CID Name, choose the FreePBX report, not the Elastix one.

You can also log in to the CLI and see what's happening when a call is coming to debug (refer to Elastix Without Tears for CLI access)
 

fenixdemetal

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#11
I already have all setings like you said, but It does not work for me.
I see the cdr report on free pbx and still shows numbers not names.

Any idea ? thanks again, for your help!
 

Patrick_elx

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#12
Patrick_elx said:
You can also log in to the CLI and see what's happening when a call is coming to debug (refer to Elastix Without Tears for CLI access)
 

fenixdemetal

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#13
ok, i see the log and I Found this in the incoming calls:

Executing [131@from-did-direct:1] GotoIf("SIP/740-b7a0a438", "0?ext-local|131|1" ) in new stack
-- Executing [131@from-did-direct:2] Macro("SIP/740-b7a0a438", "user-callerid|" ) in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/740-b7a0a438", "AMPUSER=740" ) in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/740-b7a0a438", "0?report" ) in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/740-b7a0a438", "1|Set|REALCALLERIDNUM=740" ) in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/740-b7a0a438", "AMPUSER=" ) in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/740-b7a0a438", "AMPUSERCIDNAME=" ) in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/740-b7a0a438", "1?report" ) in new stack
I realize that the values AMPUSER and AMPUSERCIDNAME are empty, but when i see in the log a internal ext. call I can see the name in the AMPUSERCIDNAME field.

do that give you any idea ? 'couse I really dont know what that means.
 

Patrick_elx

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#14
Why is your call going through the from-did-direct context? How did you call your PBX?
Do you have a proper inbound route created?
Why can't we see this route selected?

Is that all the CLI lines from the begining of the call until the macro-user-callerid?

You should have a call to a cidlookup_n context somewhere before that.
If not that means that your route is not calling the cidlookup.
 

fenixdemetal

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#15
I think this is the complete call log. from number 6477134 to ext 131.

-- Executing [s@app-announcement-1:1] GotoIf("SIP/740-b7ee33a0", "0?begin") in new stack
-- Executing [s@app-announcement-1:2] Answer("SIP/740-b7ee33a0", "") in new stack
-- Executing [s@app-announcement-1:3] Wait("SIP/740-b7ee33a0", "1") in new stack
-- Executing [s@app-announcement-1:4] NoOp("SIP/740-b7ee33a0", "Playing announcement Bienvenidos") in new stack
-- Executing [s@app-announcement-1:5] BackGround("SIP/740-b7ee33a0", "custom/Bienvenidossolo|nm") in new stack
-- <SIP/740-b7ee33a0> Playing 'custom/Bienvenidossolo' (language 'en')
-- Executing [s@app-announcement-1:6] Goto("SIP/740-b7ee33a0", "app-announcement-9|s|1") in new stack
-- Goto (app-announcement-9,s,1)
-- Executing [s@app-announcement-9:1] GotoIf("SIP/740-b7ee33a0", "1?begin") in new stack
-- Goto (app-announcement-9,s,4)
-- Executing [s@app-announcement-9:4] NoOp("SIP/740-b7ee33a0", "Playing announcement evaluar") in new stack
-- Executing [s@app-announcement-9:5] Playback("SIP/740-b7ee33a0", "custom/Evaluarservicio|noanswer") in new stack
-- <SIP/740-b7ee33a0> Playing 'custom/Evaluarservicio' (language 'en')
-- Executing [s@app-announcement-9:6] Goto("SIP/740-b7ee33a0", "ivr-2|s|1") in new stack
-- Goto (ivr-2,s,1)
-- Executing [s@ivr-2:1] Set("SIP/740-b7ee33a0", "MSG=custom/Opciones-marcado") in new stack
-- Executing [s@ivr-2:2] Set("SIP/740-b7ee33a0", "LOOPCOUNT=0") in new stack
-- Executing [s@ivr-2:3] Set("SIP/740-b7ee33a0", "__DIR-CONTEXT=default") in new stack
-- Executing [s@ivr-2:4] Set("SIP/740-b7ee33a0", "_IVR_CONTEXT_ivr-2=") in new stack
-- Executing [s@ivr-2:5] Set("SIP/740-b7ee33a0", "_IVR_CONTEXT=ivr-2") in new stack
-- Executing [s@ivr-2:6] GotoIf("SIP/740-b7ee33a0", "1?begin") in new stack
-- Goto (ivr-2,s,9)
-- Executing [s@ivr-2:9] Set("SIP/740-b7ee33a0", "TIMEOUT(digit)=3") in new stack
-- Digit timeout set to 3
-- Executing [s@ivr-2:10] Set("SIP/740-b7ee33a0", "TIMEOUT(response)=5") in new stack
-- Response timeout set to 5
-- Executing [s@ivr-2:11] Set("SIP/740-b7ee33a0", "__IVR_RETVM=") in new stack
-- Executing [s@ivr-2:12] ExecIf("SIP/740-b7ee33a0", "1|Background|custom/Opciones-marcado") in new stack
-- <SIP/740-b7ee33a0> Playing 'custom/Opciones-marcado' (language 'en')
== CDR updated on SIP/740-b7ee33a0
-- Executing [131@ivr-2:1] ExecIf("SIP/740-b7ee33a0", "0|dbDel|") in new stack
-- Executing [131@ivr-2:2] Set("SIP/740-b7ee33a0", "__NODEST=") in new stack
-- Executing [131@ivr-2:3] Goto("SIP/740-b7ee33a0", "from-did-direct|131|1") in new stack
-- Goto (from-did-direct,131,1)
-- Executing [131@from-did-direct:1] GotoIf("SIP/740-b7ee33a0", "0?ext-local|131|1") in new stack
-- Executing [131@from-did-direct:2] Macro("SIP/740-b7ee33a0", "user-callerid|") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/740-b7ee33a0", "AMPUSER=740") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/740-b7ee33a0", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/740-b7ee33a0", "1|Set|REALCALLERIDNUM=740") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/740-b7ee33a0", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/740-b7ee33a0", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/740-b7ee33a0", "1?report") in new stack
-- Goto (macro-user-callerid,s,10)
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/740-b7ee33a0", "0?continue") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/740-b7ee33a0", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/740-b7ee33a0", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("SIP/740-b7ee33a0", "Using CallerID "6477134" <740>") in new stack
-- Executing [131@from-did-direct:3] GotoIf("SIP/740-b7ee33a0", "1?skipdb") in new stack
-- Goto (from-did-direct,131,5)
-- Executing [131@from-did-direct:5] Set("SIP/740-b7ee33a0", "__NODEST=") in new stack
-- Executing [131@from-did-direct:6] Set("SIP/740-b7ee33a0", "__BLKVM_OVERRIDE=BLKVM/131/SIP/740-b7ee33a0") in new stack
-- Executing [131@from-did-direct:7] Set("SIP/740-b7ee33a0", "__BLKVM_BASE=131") in new stack
-- Executing [131@from-did-direct:8] Set("SIP/740-b7ee33a0", "DB(BLKVM/131/SIP/740-b7ee33a0)=TRUE") in new stack
-- Executing [131@from-did-direct:9] Set("SIP/740-b7ee33a0", "RRNODEST=") in new stack
-- Executing [131@from-did-direct:10] Set("SIP/740-b7ee33a0", "__NODEST=131") in new stack
-- Executing [131@from-did-direct:11] Set("SIP/740-b7ee33a0", "RecordMethod=Group") in new stack
-- Executing [131@from-did-direct:12] Macro("SIP/740-b7ee33a0", "record-enable|131|Group") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/740-b7ee33a0", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/740-b7ee33a0", "recordingcheck|20091209-143349|1260387215.3546") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20091209-143349|1260387215.3546: Recording enable for 131
recordingcheck|20091209-143349|1260387215.3546: CALLFILENAME=g131-20091209-143349-1260387215.3546
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:999] MixMonitor("SIP/740-b7ee33a0", "g131-20091209-143349-1260387215.3546.wav||") in new stack
-- Executing [131@from-did-direct:13] Set("SIP/740-b7ee33a0", "RingGroupMethod=ringallv2") in new stack
-- Executing [131@from-did-direct:14] Set("SIP/740-b7ee33a0", "_FMGRP=131") in new stack
-- Executing [131@from-did-direct:15] GotoIf("SIP/740-b7ee33a0", "0?doconfirm") in new stack
== Begin MixMonitor Recording SIP/740-b7ee33a0
-- Executing [131@from-did-direct:16] Macro("SIP/740-b7ee33a0", "dial|20|tr|131") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/740-b7ee33a0", "0?dial") in new stack
-- Executing [s@macro-dial:2] SetMusicOnHold("SIP/740-b7ee33a0", "Navidad") in new stack
-- Executing [s@macro-dial:3] AGI("SIP/740-b7ee33a0", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is '6477134' number is '740'
dialparties.agi: Methodology of ring is 'ringallv2'
-- dialparties.agi: Added extension 131 to extension map
-- dialparties.agi: Extension 131 cf is disabled
-- dialparties.agi: Extension 131 do not disturb is disabled
dialparties.agi: ExtensionState: 0
dialparties.agi: Extension 131 has ExtensionState: 0
-- dialparties.agi: Checking CW and CFB status for extension 131
-- dialparties.agi: dbset CALLTRACE/131 to 740
-- dialparties.agi: Filtered ARG3: 131
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/740-b7ee33a0", "SIP/131|22|trM(auto-blkvm)") in new stack
-- Called 131
-- SIP/131-08603d98 is ringing
I will apreciate if you take a look of it.
 

Patrick_elx

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#16
No you are still missing the beginning of the call...

What you show starts at the announcement.

There should be some stuff before going there...
 

fenixdemetal

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#17
Some word to look for ? or something to help me to find the correct part of the log ? is there any file that save that log ?
 

fenixdemetal

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#18
you mean this?:

- Executing [0@from-pstn:1] NoOp("SIP/740-087489f0", "Catch-All DID Match - Found 0 - You probably want a DID for this.") in new stack
-- Executing [0@from-pstn:2] Goto("SIP/740-087489f0", "ext-did|s|1") in new stack
-- Goto (ext-did,s,1)
-- Executing [s@ext-did:1] Set("SIP/740-087489f0", "__FROM_DID=s") in new stack
-- Executing [s@ext-did:2] Gosub("SIP/740-087489f0", "cidlookup|cidlookup_1|1") in new stack
-- Executing [cidlookup_1@cidlookup:1] LookupCIDName("SIP/740-087489f0", "") in new stack
-- Executing [cidlookup_1@cidlookup:2] Return("SIP/740-087489f0", "") in new stack
-- Executing [s@ext-did:3] ExecIf("SIP/740-087489f0", "0 |Set|CALLERID(name)=740") in new stack
-- Executing [s@ext-did:4] SetMusicOnHold("SIP/740-087489f0", "Navidad") in new stack
-- Executing [s@ext-did:5] Set("SIP/740-087489f0", "__MOHCLASS=Navidad") in new stack
-- Executing [s@ext-did:6] Set("SIP/740-087489f0", "FAX_RX=system") in new stack
-- Executing [s@ext-did:7] Set("SIP/740-087489f0", "FAX_RX_EMAIL=fenixdemetal@gmail.com") in new stack
-- Executing [s@ext-did:8] Set("SIP/740-087489f0", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [s@ext-did:9] SetCallerPres("SIP/740-087489f0", "allowed_not_screened") in new stack
-- Executing [s@ext-did:10] Goto("SIP/740-087489f0", "app-announcement-1|s|1") in new stack
-- Goto (app-announcement-1,s,1)
 

Patrick_elx

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#19
fenixdemetal said:
- Executing [0@from-pstn:1] NoOp("SIP/740-087489f0", "Catch-All DID Match - Found 0 - You probably want a DID for this." in new stack
-- Executing [0@from-pstn:2] Goto("SIP/740-087489f0", "ext-did|s|1" in new stack
-- Goto (ext-did,s,1)
-- Executing [s@ext-did:1] Set("SIP/740-087489f0", "__FROM_DID=s" in new stack
-- Executing [s@ext-did:2] Gosub("SIP/740-087489f0", "cidlookup|cidlookup_1|1" in new stack
-- Executing [cidlookup_1@cidlookup:1] LookupCIDName("SIP/740-087489f0", "" in new stack
-- Executing [cidlookup_1@cidlookup:2] Return("SIP/740-087489f0", "" in new stack
That's what I was looking for.
Also it seems that's you catch all route that took the call.
Go in your inbound route, check your catch all route (ANY/ANY) and make sure that you put the CID lookup source to CID Superfecta, because right now it's going to the CID source cidlookup_1, and I'll bet that your cidlookup_1 in extensions_additional.conf is not the one that goes to superfecta.
 

fenixdemetal

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#20
hi!, I check it and it is correct, the CID Lookup Source in my only inbound route (any DID / any CID ) is Caller ID Superfecta.

what can i do now ?
 

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